Configuration for Interoperability of Avaya Aura® Session

WDC/VG; Reviewed: SPOC 10/14/2011

Solution & Interoperability Test Lab Application Notes ©2011 Avaya Inc. All Rights Reserved.

of 125Aura61_Codian_2

Avaya Solution & Interoperability Test Lab

Configuration for Interoperability of Avaya Aura® Session Manager 6.1 and Cisco TelePresence MCU 4501 via SIP Trunk and Supporting Cisco H.323 Connected Video Endpoints - Issue 1.1

Abstract

These Application Notes describe the configuration steps necessary to connect Avaya Aura®

Session Manager 6.1 and Cisco TelePresence MCU 4501 via SIP Trunk, supporting Cisco H.323 connected video endpoints registered to the Cisco VCS. Testing was conducted at the Avaya Solution and Interoperability Test Lab at the request of the Avaya Product Management Team.



of 125 Aura61_Codian_2

Table of Contents 1. Introduction ...................................................................................................................... 5

1.1. Avaya Aura® Session Manager .........................................................................................6

1.2. Avaya Aura® System Manager .........................................................................................6

1.3. Avaya Aura® Communication Manager as an Evolution Server ......................................6

1.4. Cisco VCS .........................................................................................................................7

1.5. Cisco TelePresence MCU 4501 ........................................................................................7

2. Interoperability Testing .................................................................................................... 8

2.1. Configuration ....................................................................................................................8

2.1.1. IP Addressing ............................................................................................................ 9

2.1.2. Dial Plan.................................................................................................................. 10

2.1.3. Equipment and Software Validated ........................................................................ 11

3. Configuration ................................................................................................................. 12

3.1. Administer Avaya Aura® System Manager ....................................................................13

3.1.1. Access Avaya Aura® System Manager ................................................................... 14

3.1.2. Administer SIP Domain .......................................................................................... 15

3.1.3. Add Location .......................................................................................................... 15

3.1.4. Administer Avaya Aura® Session Manager SIP Entity .......................................... 17

3.1.5. Administer Communication Manager Evolution Server SIP Entity ....................... 20

3.1.6. Administer Cisco TelePresence MCU 4501 SIP Entity ......................................... 21

3.1.7. Administer SIP Entity Links ................................................................................... 22

3.1.8. Define Routing Policies .......................................................................................... 23

3.1.9. Define Dial Patterns ................................................................................................ 25

3.1.10. Administer Avaya Aura® Session Manager ........................................................ 28

3.1.11. Administer Avaya Aura® Communication Manager .......................................... 29

3.1.12. Add SIP Users ..................................................................................................... 36

3.2. Avaya Aura® Communication Manager Evolution Server .............................................41

3.2.1. Verify System Capabilities and Communication Manager Licensing .................... 41

3.2.2. Administer IP Node-Name ..................................................................................... 43

3.2.3. Administer IP Codec Type ...................................................................................... 44

3.2.4. Configure IP Network Region ................................................................................ 45

3.2.5. Administer Signaling Group ................................................................................... 46




3.2.6. Add SIP Trunk Group ............................................................................................. 48

3.2.7. Administer Dial Plan Analysis................................................................................ 49

3.2.8. Check Station Endpoints ......................................................................................... 50

3.2.9. Check Off PBX Endpoint Mapping ........................................................................ 52

3.2.10. Save Translations ................................................................................................ 53

3.3. Administer Avaya 10x0 Video Endpoints ......................................................................54

3.3.1. Log In Avaya 1010 Video Endpoint ....................................................................... 54

3.3.2. Set Network Preferences ......................................................................................... 55

3.3.3. Configure Communications .................................................................................... 56

3.3.4. Configure System.................................................................................................... 58

3.4. Administer Avaya one-X® Communicator .....................................................................60

3.4.1. General Settings ...................................................................................................... 60

3.5. Administer Avaya Desktop Video Device (ADVD) A175 .............................................63

3.5.1. Enable G.722 Codec on ADVD .............................................................................. 71

3.6. Administer Cisco VCS ....................................................................................................72

3.6.1. Continuing Configuration Using Web Interface ..................................................... 73

3.6.2. VCS Configuration ................................................................................................. 73

3.7. Administer Cisco TelePresence MCU 4501 ...................................................................85

3.7.1. Continue Configuration Using Web Interface ........................................................ 86

3.7.2. Administer Network Services ................................................................................. 87

3.7.3. Administer Settings ................................................................................................. 88

3.7.4. Administer Endpoints ........................................................................................... 100

3.7.5. Administer Conferences ........................................................................................ 104

3.8. Administer Cisco C20 Video Endpoint .........................................................................109

3.8.1. Continuing Configuration Using Web Interface ................................................... 109

3.8.2. Administering H.323 ............................................................................................. 110

3.8.3. Administer Network Services ............................................................................... 111

3.9. Administer Cisco 1700 MXP Video Endpoint ..............................................................113

3.9.1. Continuing Configuration Using Web Interface ................................................... 113

3.9.2. Administer System Configuration ........................................................................ 114

3.10. Administer Cisco EX-90 Video Endpoint .....................................................................115

3.10.1. Continuing Configuration Using Web Interface ............................................... 115

3.10.2. Administering H.323 ......................................................................................... 116




3.10.3. Administer Network Services ........................................................................... 117

4. Verification .................................................................................................................. 119

4.1. Avaya HQ ......................................................................................................................120

4.1.1. Registered Users on Avaya Aura® Session Manager ........................................... 120

4.1.2. Avaya Aura® Session Manager Entity Link Status .............................................. 120

4.2. Cisco Remote Branch / B2B Partner .............................................................................121

4.2.1. Registered Users on VCS...................................................................................... 121

4.2.2. MCU 4501 Conference Server.............................................................................. 121

5. Conclusion ................................................................................................................... 122

6. References .................................................................................................................... 123




1. Introduction These application notes describe the configuration necessary to connect Avaya Aura® 6.1 solution, representing a Headquarters (HQ) location with a remote branch office or Business to Business Partner (B2BP) having a Cisco video conference solution supporting multi-point conference calls using the Cisco TelePresence MCU 4501 Media Conferencing Multipoint Control Unit (MCU). See Figure 1. The connection between the two sites will utilized their WAN connections, establishing a SIP trunk between the Avaya Aura® Session Manager and the Cisco TelePresence MCU 4501. The headquarters location has Avaya 10x0 series conference endpoints registered to the Session Manager using the SIP protocol. Avaya one-X® Communicator, supporting video, is installed on Microsoft Windows stations. The Avaya one-X® Communicator supports both SIP and H.323 configurations and is registered to Session Manager and Communication Manager Evolution Server respectively depending on the configuration selected. The headquarters location has the following video endpoints:

Avaya 1010 Avaya 1020 Avaya 1030 Avaya 1040 Avaya 1050 one-X® Communicator (Windows/H.323) one-X® Communicator (Windows/SIP)

The branch/B2BP location has Cisco video endpoints consisting of the following:

Quick Set C20 1700 MXP EX90

Each of these endpoints is configured to use the H.323 protocol and use the Cisco Video Communication Server (VCS) for registration, functioning as their H.323 gatekeeper. The VCS has a H.323 trunk established between itself and the MCU 4501, providing communication between the two devices. These Application Notes will focus on the configuration of the SIP trunks between Session Manager and Communication Manager Evolution Server and between Session Manager and the Cisco TelePresence MCU 4501 . Additionally, configuration of the VCS for registration of the remote branch H.323 video endpoints and H.323 trunk between the VCS and the MCU 4501 will be covered in these application notes. Detailed administration of other aspects of Session Manager, Communication Manager, Cisco VCS, MCU 4501, and video endpoints will not be




described. For more information on these other administration actions, see the appropriate documentation listed in Section 6.

1.1. Avaya Aura® Session Manager Avaya Aura® Session Manager is a SIP routing and integration tool and the core component within the Avaya Aura® Enterprise Edition solution. It integrates all the SIP entities across the entire enterprise network within a company. Avaya Aura® Session Manager offers a new perspective on enterprise communication where individual locations are no longer managed as separate units within the enterprise. Each location, branch, or application is part of the overall enterprise, managed as an enterprise, and seen as an enterprise. Avaya Aura® Session Manager offers a simplified network-wide feature deployment; centralized routing, SIP trunking, and user profiles; cost-effective scalability from small to very large deployments, high availability with geographic redundancy and a secure environment that conforms to specific SIP standards and practices.

1.2. Avaya Aura® System Manager Central management of Avaya Aura® Session Manager is handled through the Avaya Aura® System Manager application. Avaya Aura® System Manager delivers a set of shared, secure management services and a common console across multiple products. Avaya Aura® System Manager includes the following central management services.

User Management allows for the administration of users and user groups. Communication System Management allows for the administration of individual and group stations and mailboxes.

Network Routing Policy is used for the administration of routing policy for all Avaya Aura® Session Manager instances within an Enterprise.

Alarm Management Service supports alarm monitoring, acknowledgement, configuration, clearing, and retiring.

The Logging Service receives log events formatted in the common log format. The Avaya Aura® System Manager provides miscellaneous management functions for Session Manager elements, including administering instances, configuring SIP firewalls, sequencing applications, monitoring SIP entities and security module, and managing bandwidth usage. A central database that resides on the System Manager server stores all Avaya Aura® System Manager central data, Avaya Aura® Session Manager administration data, and the Central Data Distribution Service information. The last item is used to detect changes to the Avaya Aura® System Manager central database and then distribute these changes to the Avaya Aura® Session Manager instances.

1.3. Avaya Aura® Communication Manager as an Evolution Server Avaya Aura® Communication Manager as an evolution server provides Communication Manager features to both SIP and non-SIP endpoints. It uses the full call model with Communication Manager as the only supported application. Communication Manager is administered as an evolution server by disabling the IMS-enabled? Field on the signaling group form.




With an evolution server:

H.323, digital, and analog endpoints register with Communication Manager

SIP endpoints register with Session Manager

All endpoints receive services from Communication Manager

The connection from the evolution server to the Session Manager is a non-IMS signaling group. The Session Manager routes calls from and to SIP endpoints. The SIP endpoints can then communicate with all other endpoints that are connected to the Communication Manager.

1.4. Cisco VCS The Cisco Video Communications Server (VCS) provides internal video control and administration for Cisco SIP and H.323 video devices. The VCS operates as a SIP registrar for SIP video endpoints and as a H.323 gatekeeper for H.323 video endpoints. The VCS provides H.323 ↔ SIP interworking. The VCS can also support both SIP and H.323 trunks to other devices, such the MCU 4501, providing local routing to the conference bridge.

1.5. Cisco TelePresence MCU 4501 The MCU 4501 is an entry-level, high definition (HD) video conferencing bridge. It supports 1080p HD multipoint video conferencing. The MCU 4501 is selectable between six HD and 12 SD video ports. It provides custom layouts for video conference formats. Administration of the MCU 4501 is provided through a web based interface.




2. Interoperability Testing The interoperability testing focused on the following areas:

The ability to dial into the conference bridge (MCU 4501) and create a conference by setting up the conference id and password.

Additional parties join the video conference and both video and audio are present for all conference participants.

Active screen change and indication of the active participant speaking while in a multi-party conference.

Ability to “Audio Mute” and “Audio Unmute” from video endpoints while in a multi-party conference.

Ability to “Video Mute” and “Video Unmute” from video endpoints while in a multi-party conference. Note: Not all video endpoints support this feature.

Dial out of conference to video endpoints to join multi-party conference. DTMF is recognized within conference to allow conference management and select

options. Conference participant drops from conference and is able to rejoin a multi-party

conference. Presenter is able to share presentation and video to other members in the conference.

2.1. Configuration

Figure 1: Sample Configuration




2.1.1. IP Addressing The following tables provide the IP address settings used to configure the specified devices and endpoints in the sample configuration used during testing. All devices and endpoints in this sample configuration use static IP address assignments. This was done for testing purposes and to allow monitoring and trace collection as needed. This may differ from production deployments where DHCP may be used to hand out IP address configurations. Please check with your network administrator for details on DHCP addressing assignments.

Avaya (HQ) Subnet: 192.168.230.xxx

Device IP Mask Gateway DNS CM-ES: S8300D 192.168.230.117 255.255.255.0 192.168.230.254 192.168.1.2 CM-ES: G450 192.168.230.119 255.255.255.0 192.168.230.254 192.168.1.2 Session Manager: Admin 192.168.230.123 255.255.255.0 192.168.230.254 192.168.1.2 Session Manager: SIP 192.168.230.124 255.255.255.0 192.168.230.254 192.168.1.2 System Manager 192.168.230.122 255.255.255.0 192.168.230.254 192.168.1.2 Avaya 1010 192.168.230.138 255.255.255.0 192.168.230.254 192.168.1.2 Avaya 1020 192.168.230.139 255.255.255.0 192.168.230.254 192.168.1.2 Avaya 1030 192.168.230.140 255.255.255.0 192.168.230.254 192.168.1.2 Avaya 1040 192.168.230.141 255.255.255.0 192.168.230.254 192.168.1.2 Avaya 1050 192.168.230.142 255.255.255.0 192.168.230.254 192.168.1.2 Avaya ADVD #1 (A175) 192.168.230.132 255.255.255.0 192.168.230.254 192.168.1.2 Avaya ADVD #2 (A175) 192.168.230.133 255.255.255.0 192.168.230.254 192.168.1.2 Avaya 1XC-Win (SIP) 192.168.230.135 255.255.255.0 192.168.230.254 192.168.1.2 Avaya 1XC-Win (H.323) 192.168.230.136 255.255.255.0 192.168.230.254 192.168.1.2

Table 1: Avaya (HQ) IP Address Assignment

Cisco (Remote Branch/B2B Partner) Subnet: 192.168.231.xxx

Device IP Mask Gateway DNS VCS 192.168.231.51 255.255.255.0 192.168.231.254 192.168.1.2 MCU 4501 192.168.231.52 255.255.255.0 192.168.231.254 192.168.1.2 Cisco C20-1 192.168.231.60 255.255.255.0 192.168.231.254 192.168.1.2 Cisco C20-2 192.168.231.61 255.255.255.0 192.168.231.254 192.168.1.2 Cisco 1700 MXP 192.168.231.65 255.255.255.0 192.168.231.254 192.168.1.2 Cisco EX90 192.168.231.66 255.255.255.0 192.168.231.254 192.168.1.2

Table 2: Cisco (Remote Branch/B2B Partner) IP Address Assignment




2.1.2. Dial Plan The dial plan used for calling between locations and into conference bridges is show in the table below:

Avaya Video Endpoints (HQ)

Video Endpoint – Line #

Ext. Protocol Registration

Name

First Last Login

@tandberg.avaya.com

Avaya 1010 – 1 60000 SIP SM 60000(CISCO_SM) AV1010 60000 Avaya 1010 – 2 60001 SIP SM 60001(CISCO_SM) AV1010 60001 Avaya 1020 – 1 60002 SIP SM 60002(CISCO_SM) AV1020 60002 Avaya 1020 – 2 60003 SIP SM 60003(CISCO_SM) AV1020 60003 Avaya 1030 – 1 60004 SIP SM 60004(CISCO_SM) AV1030 60004 Avaya 1030 – 2 60005 SIP SM 60005(CISCO_SM) AV1030 60005 Avaya 1040 – 1 60006 SIP SM 60006(CISCO_SM) AV1040 60006 Avaya 1040 – 2 60007 SIP SM 60007(CISCO_SM) AV1040 60007 Avaya 1050 – 1 60008 SIP SM 60008(CISCO_SM) AV1050 60008 Avaya 1050 – 2 60009 SIP SM 60009(CISCO_SM) AV1050 60009 Avaya ADVD – 1 60010 SIP SM 60010(CISCO_SM) A175 60010 Avaya ADVD – 2 60011 SIP SM 60011(CISCO_SM) A175 60011 Avaya 1XC – Win 60050 SIP SM 60050(CISCO_SM) 1XC-SIP 60050 Avaya 1XC – Win 61000 H.323 CM 61000(CISCO_SM) 1XC-H323 61000

Table 3: Avaya HQ Dial Plan

Cisco Video Endpoints (Remote Branch/B2B Partner)

Video Endpoint Ext. Protocol Registration Name Cisco C20 65000 H.323 VCS 65000 Cisco C20 65001 H.323 VCS 65001

Cisco 1700 MXP 65005 H.323 VCS 65005 Cisco EX90 65010 H.323 VCS 65010

Cisco TelePresence MCU 4501

69500 H.323 VCS 69500

Table 4: Cisco Remote Branch Dial Plan




2.1.3. Equipment and Software Validated The following equipment and software were used for the sample configuration provided: Provider Hardware Component Software Version

Avaya S8300D with G450 Media Gateway Avaya Aura® Communications Manager acting as an Evolution Server Version: 00.1.510.1-19080

Avaya S8800 Media Server Session Manager 6.1 Avaya S8800 Media Server System Manager 6.1 Cisco Video Conference Server Version: X6.0 Build: 211839 Cisco MCU 4501 Version: 4.2(1.43) Build: 6.17(1.43) Avaya 1010 Video Conferencing Endpoint Version: 4.7.3(14) Avaya 1020 Video Conferencing Endpoint Version: 4.7.3(14) Avaya 1030 Video Conferencing Endpoint Version: 4.7.3(14) Avaya 1040 Video Conferencing Endpoint Version: 4.7.3(14) Avaya 1050 Video Conferencing Endpoint Version: 4.7.3(14)

Avaya Avaya one-X® Communicator (Windows/H.323)

Version: 6.1.1.02-SP1-32858

Avaya Avaya one-X® Communicator (Windows/SIP) Version: 6.1.1.02-SP1-32858

Avaya Avaya Flare™ Experience / Avaya Desktop Video Device

Version: 1.0.3

Cisco EX-90 Desktop Video System Version: TC4.1.0.247017 Cisco 1700 MXP Desktop Video System Version: F9.0.2 NTSC Cisco Quick Set C20 Version: TC4.0.1.240265




3. Configuration The sample configuration used in these Application Notes will focus on configuring:

Avaya Headquarters o Avaya Aura® Session Manager o Avaya Aura® Communication Manager acting as an Evolution Server o Avaya 10x0 Video Endpoints o Avaya one-X® Communicator o Avaya Desktop Video Device (ADVD) A175

Cisco Remote Branch/B2BP

o Cisco VCS o Cisco TelePresence MCU 4501 o Cisco C20 Video Endpoints o Cisco 1700 MXP Video Endpoint o Cisco EX-90 Video Endpoint




3.1. Administer Avaya Aura® System Manager The following section describes administering SIP Entities between Session Manager and the Communication Manager Evolution Server in order to establish a SIP Entity link between Session Manager and the Communication Manager Evolution Server and a SIP Entity link between Session Manager and the MCU 4501 at the remote branch location.

Access Avaya Aura® System Manager Add SIP Domain Add Location Administer Avaya Aura® Session Manager SIP Entity Administer Avaya Aura® Communication Manager Evolution Server SIP Entity Administer Cisco TelePresence MCU 4501 SIP Entity Administer SIP Entity Links Define Routing Policies, which control call routing between the SIP Entities. Define Dial Patterns, which govern to which SIP Entity a call is routed. Administer Avaya Aura® Session Manager Administer Avaya Aura® Communication Manager Add SIP User’s




3.1.1. Access Avaya Aura® System Manager Access the System Manager web interface, by entering http://<ip-addr>/SMGR as the URL in an Internet browser, where <ip-addr> is the IP address of the server running System Manager graphical user interface. Log in with the appropriate Username and Password and press the Log On button to access System Manager.

The Home Menu Tab of the System Manager Graphical User Interface is displayed in the following screenshot.




3.1.2. Administer SIP Domain From the Home Menu Tab, under the Elements section, select Routing → Domains and click on the New button to add a new domain.

Enter the following values and use default values for remaining fields.

Name Enter the Authoritative Domain Name. In the sample configuration, “tandberg.avaya.com” was used.

Type Verify “SIP” is selected. Notes Add a brief description. [Optional]

Click Commit to save. The screen below shows the SIP Domain defined for the tandberg.avaya.com domain.

3.1.3. Add Location Locations are used to identify logical and physical locations where SIP entities reside for the purposes of bandwidth management or location based routing.




To add a new Location, click on Routing → Locations and click on the New button.

In the General section, enter the following values and use default values for remaining fields.

Name: Enter a descriptive name for the location. For the sample configuration, “VIDEO_LOCATION” was used.

Notes: Add a brief description. [Optional]

In the Per-Call Bandwidth Parameters section, enter the following value. Default Audio

Bandwidth: The default values of “80” Kbit/sec was used. Click Commit to save. The screen below shows the Location defined for the sample configuration.




3.1.4. Administer Avaya Aura® Session Manager SIP Entity A SIP Entity must be added for each video/telephony system connected to Session Manager over SIP trunk(s). Create a SIP Entity for the Session Manager by selecting Routing → SIP Entities and then click on the New button.




In the General section, enter the following values and use default values for remaining fields.

Name: Enter an identifier for the SIP Entity. “CISCO_SM” was used for the sample configuration.

FQDN or IP Address: Enter IP Address of Session Manager: SIP address. Refer to Table 1.

Type: Select “Session Manager”. Notes: Enter a brief description. [Optional] Location: Select the Location Defined for Session Manager.

“VIDEO_LOCATION” was used for the sample configuration. Time Zone: Select the appropriate time zone for Session Manager.

In the SIP Link Monitoring section: SIP Link

Monitoring: Select “Use Session Manager Configuration”

In the Port section, click the Add button and populate with the following values:

Port “5060” Protocol “TCP” Default Domain Select domain defined in Section 3.1.2

In this sample configuration,“tandberg.avaya.com” is used Notes Enter a brief description. [Optional]




Click the Add button to create another entry and populate with the following values: Port “5060” Protocol “UDP” Default Domain “tandberg.avaya.com” Notes Enter a brief description. [Optional]

Once again, click the Add button to create another entry and populate with the following values:

Port “5061” Protocol “TLS” Default Domain “tandberg.avaya.com” Notes Enter a brief description. [Optional]

Click the Commit button to save the definition of the new SIP Entity. The following screen shows the port setting configured for this sample configuration.

Click the Commit button to confirm changes.




3.1.5. Administer Communication Manager Evolution Server SIP Entity The CM-ES SIP Entity is the second part of the link between the Session Manager and the Communication Manager Evolution Server. Create a SIP Entity for the CM-ES by selecting Routing → SIP Entities and then click on the New button. In the General section, enter the following values and use default values for remaining fields.

Name: Enter an identifier for the SIP Entity. “CM_ES” was used for the sample configuration.

FQDN or IP Address: Enter IP Address of S8300D card in G450 Media gateway. Refer to CM-ES: S8300D in Table 1.

Type: Select “CM”. Notes: Enter a brief description. [Optional] Location: Select the Location Defined for the CM-ES.

“VIDEO_LOCATION” was used for the sample configuration. Time Zone: Select the appropriate time zone for the CM-ES.

In the SIP Link Monitoring section:

SIP Link Monitoring: Select “Use Session Manager Configuration”.

Click the Commit button to confirm changes.




3.1.6. Administer Cisco TelePresence MCU 4501 SIP Entity The Cisco TelePresence MCU 4501 SIP Entity is created to provide the far side SIP Entity allowing a SIP Trunk connection between Session Manager and the MCU 4501. Create a SIP Entity for MCU 4501 by selecting Routing → SIP Entities and then click on the New button. In the General section, enter the following values and use default values for remain fields.

Name: Enter and identifier for the SIP Entity. “TandbergCodian” was used for the sample configuration.

FQDN or IP Address: Enter IP Address of MCU 4501. Refer to MCU 4501 in Table 2.

Type: Select “SIP Trunk”. Notes: Enter a brief description. [Optional] Location: Select the Location Defined in Section 3.1.3.

“VIDEO_LOCATION” was used for the sample configuration. Time Zone: Select the appropriate time zone for the MCU 4501.

In the SIP Link Monitoring section:

SIP Link Monitoring: Select “Use Session Manager Configuration”.

Click the Commit button to save changes.




3.1.7. Administer SIP Entity Links A SIP trunk between Session Manager and a video/telephony system is described by an Entity Link. In the sample configuration, SIP Entity Links were added between Session Manager and each telephony system. Two SIP Entity Links need to be created for this sample configuration:

1. SIP Entity Link between Session Manager and CM-ES. 2. SIP Entity Link between Session Manager and MCU 4501.

Create the SIP Entity Link by selecting Routing → Entity Links and clicking on the New button. Enter the following values:

Name Enter an identifier for the link to each video/telephony system. SIP Entity 1 Select SIP Entity defined for Session Manager in Section 3.1.4. SIP Entity 2 Select the SIP Entity defined for CM-ES in Section 3.1.5. Protocol After selecting both SIP Entities, select “TCP” as the required

protocol. Port Verify Port for both SIP entities is the default listen port.

For the sample configuration, default listen port is “5060”. Trusted Enter . Notes Enter a brief description. [Optional]

Click Commit to save the Entity Link definition. The following screen shows the entity link defined for the SIP trunk between Session Manager and Avaya Communication Manager Evolution Server.




Repeat the steps to create a SIP Entity Link for the link between the Session Manager and the MCU 4501 using the settings shown below.

3.1.8. Define Routing Policies Routing policies describe the conditions under which calls will be routed to the SIP Entities specified in Sections 3.1.4 – 3.1.6. Two routing policies are needed, one for the MCU 4501 and another for the CM-ES. Create a routing policy by selecting Routing → Routing Policies and then click on the New button. In the General section, enter the following values:

Name: Enter an identifier to define the routing policy. Disabled: Leave unchecked, . Notes: Enter a brief description. [Optional]




In the SIP Entity as Destination section, click Select. The SIP Entity Link page opens (not shown).

Select the SIP Entity associated with CM-ES defined in Section 3.1.5 and click Select.

The selected SIP Entity displays on the Routing Policy Details page. Use default values for remaining fields. Click Commit to save Routing Policy definition.




Repeat to create a routing policy to the MCU 4501. The following screen shows the routing policy created to the MCU 4501 used in the sample configuration.

3.1.9. Define Dial Patterns Define dial patterns to direct calls to the appropriate video/telephony system. In the sample configuration, 5-digit extensions beginning with “695” are directed to the MCU 4501 as the conference bridge number is configured as “69500”. All local Avaya HQ extensions will fall under a general 5-digit dial plan for numbers starting with “6”. The numbering scheme used in the sample configuration has “60xxx” extensions for Avaya SIP video endpoints and “61xxx” for Avaya H.323 video endpoints. Create a dial plan by selecting Routing → Dial Patterns and then clicking on the New button. In the General section, enter the following values and use default values for remaining fields.

Pattern: Enter dial pattern for calls to MCU 4501. In the sample configuration, “695” was used as the pattern.

Min: Enter the minimum number digits that must be dialed. Max: Enter the maximum number digits that may be dialed. SIP Domain: Select the SIP Domain from drop-down menu or select “All” if

Session Manager should accept incoming calls from all SIP domains. The sample configuration used “tandberg.avaya.com”.

Notes: Enter a brief description. [Optional] In the Originating Locations and Routing Policies section, click Add.




The Originating Locations and Routing Policy List page opens (not shown). In Originating Locations table, select “-ALL-”. In Routing Policies table, select the Routing Policy defined for the MCU 4501 in

Section 3.1.8. Click Select to save these changes and return to Dial Pattern Details page.

Click Commit to save. The following screen shows the Dial Pattern defined for the “695”.




The following screen shows the Dial Pattern defined for routing calls to all local Avaya HQ extensions, including SIP and H.323.




3.1.10. Administer Avaya Aura® Session Manager In order to provide the link between Session Manager and System Manager, Session Manager must be added to the configuration. From the Home Menu Tab, under Elements select Session Manager to open the menu for Session Manager. Select Session Manager Administration on the left hand side of the Session Manager GUI. In Session Manager Instances section, select the New button to create a new instance.

In the General section, enter the following values:

SIP Entity Name Select the SIP Entity name for the Session Manager created in Section 3.1.4.

Description Enter a brief description. Management Access

Point Host Name/IP IP Address of the Session Manager: Admin interface. Refer to Section 2.1.1, Table 1.

Direct Routing to Endpoints Select “Enable”.




In the Security Module section, complete with the following values:

SIP Entity IP Address Enter the IP Address of the Session Manager: SIP interface. Refer to Section 2.1.1, Table 1.

Network Mask Enter the Network Mask of the Session Manager: SIP interface. Refer to Section 2.1.1, Table 1.

Default Gateway Enter the Default Gateway for the Session Manager: SIP interface. Refer to Section 2.1.1, Table 1.

Click Commit to save.

3.1.11. Administer Avaya Aura® Communication Manager In order for Communication Manager to provide configuration and feature support to Avaya SIP video/telephones when they register to Session Manager, Communication Manager Evolution Server must be added as an application for Session Manager. This is a four step process.

3.1.11.1 Step 1: Create a Communication Manager Element From the Home Menu Tab, under Elements select Inventory to open the tabbed menu for Inventory. Select Manage Elements on the left. Click on New (not shown).




Under the Application tab, enter the following fields, and use defaults for the remaining fields:

Name A descriptive name. The sample configuration uses “CM_ES-Element”.

Type Select “CM” Node Enter IP address for Communication Manager SAT access.

Under the Attributes tab, enter the following fields, and use defaults for the remaining fields:

Login Enter Login used for SAT access. Password Enter Password used for SAT access. Confirm

Password Repeat Password used for SAT access. Is SSH

Connection Checked, . Port Enter “5022”




Click Commit to save. This will set up data synchronization with Communication Manager to occur periodically in the background.




3.1.11.2 Step 2: Configure Application

From the Session Manager tab, select Session Manager → Application Configuration → Applications on the left. Click on New (not shown). Enter the following fields, and use defaults for the remaining fields:

Name Enter a descriptive name. SIP Entity Select the Communication Manager SIP Entity defined in Section

3.1.5. CM System for

SIP Entity Select Communication Manager Element defined in Step 1. Description Enter a brief description. [Optional]

Click on Commit. The screen shown below is the Edit screen from the sample configuration, since the Application has already been configured.




3.1.11.3 Step 3: Configure Application Sequence

From the Session Manager menu, select Session Manager → Application Configuration → Application Sequences on the left. Click the New button (not shown). Under the Application Sequence section, enter the following fields, and use default values for all other fields.

Name Enter a descriptive name. The sample configuration uses “CM_ES-Seq”.

Description Enter a brief description. [Optional] Under the Available Applications, click on the “+” sign next to the appropriate Available Applications, and the selected available application will be moved up to the Applications in this Sequence section. In this sample configuration, “CM_ES-App” was shown in the screen below (which is the Edit screen since the Application Sequence has already been configured). Click on Commit.




3.1.11.4 Step 4: Synchronize CM Data

From the Inventory menu, select Inventory → Synchronization → Communication System on the left. Select the check box next to the appropriate Element Name (“CM_ES-Element” in this case). Check the radio button next to “Initialize data for selected devices”. Then click on the Now button to start a data synchronization task. This may take some time to complete.

Under Services on the Home page menu, select Scheduler to open the scheduler tab. Select Scheduler → Completed Jobs to determine when the task has completed, as shown below (see entry with embedded Communication Manager name “CM_ES-Element” from the sample configuration).




3.1.12. Add SIP Users Refer to Table 3 for adding SIP Users for this sample configuration. Use the listing in Table 3 and repeat the steps for each user of the remaining SIP video/phones used in this sample configuration.

3.1.12.1 Create New User From the Home menu, under the Users section, select User Management. To add a SIP User for Avaya SIP video/phone to Session Manager, access the User Management → Manage Users on the left hand side of the System Manager GUI, then click on New to open the New Users page.

3.1.12.2 Identify New User These application notes will refer to the configuration for the first user listed in Table 3, which can be used as a reference for configuring the remaining stations listed in Table 3. In the Identity section on the Identity tab, enter the following fields with the information described below and use default values for any remaining fields.

Last Name: Enter the Last Name of the user. In the sample configuration, “AV1010” was used.

First Name: Enter the First Name of the user. In the sample configuration, “60000(CISCO_SM)” was used.

Middle Name: Enter the Middle Name of the user. [Optional] Login Name: Enter extension <no.>@<sip domain>.

In the sample configuration, [email protected]” was used.




Authentication Type: Select “Basic” from the drop down list.

Password: Enter password which will be used to log into System Manager application (password). Note: This field is only displayed if adding a new user.

Confirm Password: Enter valued entered above. Note: This field is only displayed if adding a new user.

3.1.12.3 Create a New User Communication Profile Select the Communication Profile tab and under the Communication Profile section, click on the New button to define a Communication Profile for the new SIP user. Enter values for the following required attributes and use default values for the remaining fields.

Communication Profile Password: Enter a numeric value which will be used to logon to SIP video/phone.




Confirm Password: Repeat numeric password entered above. Name: Enter name of communication profile.

“Primary”, the default username, was used in the sample configuration.

Default: Enter checkmark to indicate this profile is the default profile.

Select New to define a Communication Address for the new SIP user. Enter values for the following required attributes:

Type: Select “Avaya SIP”. Fully Qualified

Address: Enter extension number. Note: value is shown in Handle field after address is added.

@ Enter SIP domain. The sample configuration uses “tandberg.avaya.com”.

Click Add (not shown) to save the Communication Address for the new SIP user.




3.1.12.4 Assign Application Sequence to Communication Profile Assign the Application Sequence defined in Section 3.1.11.3 to the new SIP user as part of defining the Communication Profile. The Application Sequence must be used for both the originating and terminating sequence. Select the Session Manager Profile box and enter the appropriate values for the following attributes:

Primary Session Manager: Select the appropriate Session Manager instance. The sample configuration uses “CISCO_SM”.

Origination Application Sequence: Enter the Application Sequence. The sample configuration uses “CM_ES-Seq”.

Termination Application Sequence: Enter the Application Sequence. The sample configuration uses “CM_ES-Seq”.

Home Location: Select the appropriate location that was created in Section 3.1.3. The Session Manager Profile section from the sample configuration is shown below.

3.1.12.5 Configure the Endpoint Profile Select the Endpoint Profile box and enter the appropriate values for the following attributes:

System: Select the Communication Manager Element defined in Section 3.1.11.1 from drop down menu.

Profile Type: Enter “Endpoint”.




Use Existing Stations: Enter a checkmark if station was already defined, else, station will automatically be created.

Extension: Enter extension number. Template: Select the template (system defined or user defined) you want to

associate with the endpoint. Select the template based on the set type you want to add.

Set Type: Select “9630SIP” for this video endpoint. Security Code: Enter numeric value which will be used to logon to SIP phone.

Note: This field must match the value entered for the Communication Profile Password field.

Port: Select “IP” from the search list. The screen below show the Endpoint Profile section used in the sample configuration.

Repeat Section 3.1.12.1 – 3.1.12.5 add the remaining Avaya SIP Users assigned to station numbers in Table 3.




3.2. Avaya Aura® Communication Manager Evolution Server This section describes the administration of Communication Manager Evolution Server using the Communication Manager section on System Manager. These instructions assume the G450 Media Gateway is already configured on the Communication Manager Evolution Server. Some administration screens have been abbreviated for clarity.

Verify System Capabilities and Communication Manager Licensing Administer IP Node-Name Administer Codec Type Administer IP Network Region Administer Signaling Group Administer Trunk Group Administer Numbering Plan Check Station Endpoints Check Off PBX Endpoint Mapping Save Translations

3.2.1. Verify System Capabilities and Communication Manager Licensing From the Home menu, under the Elements section, select Communication Manager → Parameters → System Parameters – Customer Options. Select the radio button for the platform, and select View (not shown).

3.2.1.1 Check Off-PBX Telephones – OPS License Verify that Maximum Off-PBX Telephones – OPS has been set to the value that has been licensed, and that this value will accommodate addition of the SIP video endpoints. If a required feature is not enabled or there is insufficient capacity, contact an authorized Avaya sales representative to obtain additional capacity. The sample configuration screen below shows the customer options available.




3.2.1.2 Check Maximum Video Capable IP Softphone Licenses Verify that there are sufficient licenses to administer Video Capable IP Softphones. This is the Maximum Video Capable IP Softphones value on Page 2 of System Parameter Customer-Options.

3.2.1.3 Check Maximum Administered SIP Trunk Licenses Verify there are sufficient licenses to administer the SIP Trunk. This is the Maximum Administered SIP Trunks value on Page 2 of the System-Parameters Customer-Options. The sample configuration screen below shows Page 2 of the System-Parameters Customer -Options.




3.2.2. Administer IP Node-Name From the Home menu, under Elements, select Communication Manager. Select Communication Manager → Network → Node Names. Add the node-name and IP address for the Session Manager’s software asset card, if not previously added, by selecting the New button and filling out the Name and IP Address of the Session Manager: SIP interface from Table 1.




3.2.3. Administer IP Codec Type This configuration step uses the System Access Terminal (SAT) to administer the codec selection and order on the CM-ES. From the command line inside the SAT, issue the command change ip-codec-set n where “n” is the next available number. Enter the following values:

Enter “G.711MU” and “G.729” as supported types of Audio Codec. Silence Suppression: Retain the default value “n”. Frames Per Pkt: Enter “2”. Packet Size (ms): Enter “20”. Media Encryption: Enter the value based on the system requirement. For the

sample configuration “none” was used. change ip-codec-set 1 Page 1 of 2 IP Codec Set Codec Set: 1 Audio Silence Frames Packet Codec Suppression Per Pkt Size(ms) 1: G.711MU n 2 20 2: G.729 n 2 20 3: 4: 5: 6: 7: Media Encryption 1: none 2: 3:




3.2.4. Configure IP Network Region Using the System Manager web administration and starting from the Home Menu Tab, under Elements, select Communication Manager. Select Communication Manager → Network → IP Network Regions. Select Region 1 and click the Edit button.

Set the following fields as listed below and leave all other fields at their default values:

Authoritative Domain: Enter the Authoritative Domain to use. The sample configuration uses “tandberg.avaya.com”.

Name: Enter the name for this ip-network-region. Intra-region IP-IP

Direct Audio: Enter “yes”. Inter-region IP-IP

Direct Audio: Enter “yes”. The screen below shows the settings used in the sample configuration.




3.2.5. Administer Signaling Group Select Communication Manager → Network → Signaling Groups from the navigation menu on the left. Click the New button. Select the CM-ES and enter the signaling group number in the “Enter Qualifier” field. In the sample configuration “10” was entered. Click on the Add button to enter the Signaling Group configuration screen.





Group Type: “sip” IMS Enabled? “n” Transport Method: “tcp” IP Video? “y” Priority Video? “y” Peer Detection

Enabled? “y” Peer Server: Use default value. Near-end Node Name: “procr” Far-end Node Name: Session Manager node name from Section 3.2.2. Near-end Listen Port: “5060” Far-end Listen Port: “5060” Far-end Network

Region: “1” Far-end Domain: Leave blank. Direct IP-IP

Audio Connections? “y” Enable Layer 3 Test: “y” Initial IP-IP

Direct Media? “y”

Press the Enter button to save.




3.2.6. Add SIP Trunk Group Add the corresponding trunk group controlled by this signaling group by selecting Communication Manager → Groups → Trunk Group and clicking the New button. Select the CM-ES and enter the trunk group number in the “Enter Qualifier” field. This was set to “10” in the sample configuration. Click on the Add button to enter the Trunk Group configuration page.


Group Type: “sip” Group Name: Enter a descriptive name. TAC: Enter an available trunk access code.

*10 was used in the sample configuration. Service Type: “tie” Signaling Group: The number of the signaling group added in Section 3.2.5. Number of

Members: The number of SIP trunks to be allocated to calls routed to Session Manager (must be within the limits of the total number of trunks configured).




Click the Enter button to save.

3.2.7. Administer Dial Plan Analysis This section describes the Dial Plan Analysis configuration. This is Communication Manager’s way of translating digits dialed by the user. The user can determine the beginning digits and total length for each type of call that Communication Manager needs to interpret. The Dialed String beginning with the number “6” and with a Total Length of 5 digits will be used to administer the extension range used for the SIP video/phones. From the Communication Manager tab, select Communication Manager → System → Dialplan Analysis. Click on the New button. Select CM-ES and enter the digit(s) in the “Enter Qualifier” field. The sample configuration uses “6”. Click the Add button to configure the Dialplan Analysis page.





Dialed String “6” Total Length “5” Call Type “ext”

Click the Enter button to save.

3.2.8. Check Station Endpoints The method is the same for administering all of the Avaya 1000 series video endpoints with the exception of the 1040 and 1050’s. The only difference is that the 1040 can be administered to have up to 3 call appearances and the 1050 can have up to 7 call appearances for conferencing via their internal MCU’s. The 1010, 1020, and 1030 have to be administered with only one call-appearance since they are a single-line endpoint with no conferencing or transferring capabilities. For each SIP user to be defined in Session Manager, add a corresponding station on the Communication Manager Evolution Server. Note: instead of manually defining each station using the Communication Manager SAT interface, the preferred option is to automatically generate the SIP station when adding a new SIP user. See Section 3.1.12 for more information on adding SIP users. From the Communication Manager tab, select Communication Manager → Endpoints → Manage Endpoints. Check the box next to the station extension to be checked and select the Edit button (not shown) to enter the Endpoint configuration page. Note: It is important to assign only one call-appearance for the 1010, 1020, and 1030’s.




Check the number of call-appearances by selecting the Button Assignment (B) tab from the edit page. Below the screen will refresh with the listing of the Main Buttons in a numbered list with the function listed at the side. Check that the Avaya 1010, 1020, and 1030 models only have number one assigned to “call-appr” and the rest are set to “Select”. See the screen below for the Avaya 1010 with station number “60000”.

Check all the Avaya 1000 series video endpoints for the proper configuration for call-appearance based on the specific model.




3.2.9. Check Off PBX Endpoint Mapping All endpoint stations created in Section 3.1.12 should already have the Off PBX Endpoint Mapping configured. This is just a check from the CM-ES side. To check an endpoint, select the Communication Manager tab. Select Communication Manager → Endpoints → Off PBX Endpoint Mapping. Select an extension and click the Edit button to enter the configuration page. On “Page 1”, check the following:

Application Verify “OPS” is assigned Phone Number Contains the correct phone extension Trunk Selection Is set to “aar”. Config Set “1”

On Page 2, check the following:

Mapping Mode “both” Calls Allowed “all”




3.2.10. Save Translations Configuration of the Communication Manager Evolution Server is complete. To save the recent configurations, select the Inventory tab and then select Inventory → Synchronization → Communication System. Check the box next to the CM-ES element name and click the radio button “Save Translations for selected devices”. Click the “Now” button to save the translations.




3.3. Administer Avaya 10x0 Video Endpoints The Avaya 10x0 configuration screens are very similar, and require the same basic configuration information listed for each of the devices in Table 1. Therefore, this application note will cover setting up the configuration for one of the Avaya 10x0 series endpoints, the Avaya 1010 and individual references for each of the Avaya 10x0 series models can be found in the Reference Section 6, items 8 – 12 for further information. To administer the 10x0 video endpoints log in to the web interface using the IP address of the video endpoint. You will be redirected to a screen that looks similar to the one below. This is a sample configuration on how to administer a 10x0 video endpoint.

3.3.1. Log In Avaya 1010 Video Endpoint Enter the proper login credentials and press Submit. Most of the Preferences can be customized to meet your needs. Mentioned below are the absolute necessary items that need to be administered to get the 10x0 up and running on the network.




3.3.2. Set Network Preferences Once logged in select the Preferences tab and then the Network option.

Select the General option and enter values for the following required attributes.

DHCP: Select from the drop down menu, Enabled or Disabled. Note: DHCP was used the first time the Avaya 10x0 was powered on to gain access to the web administration interface. It was changed to Disabled and assigned static addresses for testing.

IP Address: Enter the assigned static IP Address if DHCP is disabled. Refer to Table 1 for addresses used for the sample configuration.

Subnet Mask: Enter the Subnet Mask if DHCP is disabled. Refer to Table 1.

Default Gateway: Enter the Default Gateway IP Address. Refer to Table 1.

Hostname: Enter the appropriate Hostname. DNS Servers: Enter the appropriate DNS Servers. Refer to Table 1. Network Speed: Select “Auto”. NTP Server

Hostname: Enter the Network Time Protocol (NTP) Server’s IP Address. The sample configuration uses “192.168.1.2”.

Select the Save Changes button to save.




3.3.3. Configure Communications Select the Preferences tab and select the Communications option.




Select the SIP option.

SIP: Select “Enabled”. SIP Username: Enter the SIP Username for the device.

Note: The SIP Username should be unique and meaningful to the endpoint.

Authorization Name: Enter the SIP Server authorization username. Note: The Authorization Name should be unique and meaningful to the endpoint.

Authorization Password: Enter the SIP Server authorization password.

SIP Registration: Select the communication path to use when registering with a SIP Registrar

SIP Proxy: Select “Enabled” to use the SIP proxy Proxy Hostname: Enter the hostname or IP address of the SIP proxy server.

Note: This is the Session Manager software asset card IP address Proxy IP Port: Enter the IP port number of SIP proxy server SIP Registrar: Select “Enabled” to use the SIP registrar Registrar

Hostname: Enter the hostname or IP address of the SIP registrar server.




Registrar IP Port: Enter the IP port number of the SIP registrar server.

UDP Signaling Port: Enter the UDP port number of the SIP configuration.

TCP Signaling: Select “Enabled” to use TCP for placing SIP calls. TCP Signaling

Port: Enter the TCP port number of the SIP configuration. Select the Save Changes button to save the administration just added.

3.3.4. Configure System Select the Preferences tab and select System option.




Select the Identification option. This option will allow the user to display the name and video/voice numbers on the menu bar.

System Name: Enter a descriptive name for the system. Video Number: Enter the video number of the system. Voice Number: Enter the voice number of the system.

Select the Save Changes button to save the administration just added.

Repeat the configuration steps in Section 3.3 for the remaining Avaya 10x0 series video endpoints listed in Table 1.




3.4. Administer Avaya one-X® Communicator These Application Notes assume that the Avaya one-X® Communicator has been installed on a personal computer installed with a High Definition camera and the computer meets at least the minimum requirement for the Avaya one-X® Communicator. See Reference 14 in Section 6 for more information about installing the Avaya one-X® Communicator.

3.4.1. General Settings

3.4.1.1 Phone Click Phone on the left pane of the General Settings window. The Phone settings appear on the right pane of the General Settings window. This pane displays different fields depending on whether you selected H.323 or SIP protocol during installation. H.323 protocol

1. Enter the following information: a. Extension: enter the extension for this device on the Avaya server. b. Password: enter this extension’s password.

2. To specify the IP address of the server, click Add below the Server List field.

The Add Server dialog box appears (not shown).

3. In the Server field, enter the IP address of the server. This would be the IP address of CM-ES:S8300D in Table 1.

4. Check the Enable Video Calls check box.

5. Click OK. Click the OK button to save the configuration. SIP protocol

1. Enter the following information: a. Extension: enter the extension for this device on the Avaya server. b. Password: enter this extension’s password.

2. To specify the IP address of the server, click Add below the Server List field.

The Add Server dialog box appears.

3. In the Proxy Server field, enter the IP address of the proxy server. This would be the IP address of the Session Manager: SIP in Table 1.




4. (Optional) In the Port field, enter the port number of the server. If the port is not specified, Avaya one-X® Communicator uses the default 5061 port.

5. Enter the following information: a. In the Domain field, enter the domain of the Session Manager.

b. In the Failback Policy field, select Auto to make Avaya one-X® Communicator

automatically recover after failback. Select Admin if you want Avaya one-X® Communicator to use the failback policy defined by the system administrator.

c. In the Registration Policy field, select a registration policy to specify how your extension manages proxies.

6. Check the box next to Enable Video Calls.

Click the OK button to save the configuration.

3.4.1.2 Advanced Click Advanced on the left pane of the General Settings window. The Advanced settings appear on the right pane of the General Settings window. Under the Network Settings section:

Specify local UDP/TCP port range preference Checked, .




Low Enter the number of the lowest port to use. Default is “1024”.

High Enter the number of the highest port to use. Default is “65535”.

Click the OK button to save the configuration.




3.5. Administer Avaya Desktop Video Device (ADVD) A175 This section describes steps needed to configure and connect the Avaya Desktop Video Device to Session Manager. It will explain the configuration steps needed to register the Avaya Desktop Video Device to Session Manager with Communication Manager running as an Evolution Server. It will explain how to assign an IP Address to the Avaya Desktop Video Device for access through the LAN. On applying power to the Avaya Desktop Video Device, the following screenshot is shown. Click on the Settings icon on the left hand side of the Avaya Desktop Video Device. The Settings menu appears with a number of options to configure the device. Click on the Network and Wireless option.




Within the Network and Wireless Settings menu is a further configuration menu. Click on the option for Ethernet Settings.




Within the Ethernet Settings menu is a subheading called Ethernet Settings. Click on Ethernet Settings as this is where the Ethernet interface is configured.




The Configure Ethernet Device page is displayed. Enter the field values below, refering to Table 1 for IP network address settings of the ADVD devices.

Static IP Radio button selected for Static IP. IP Address Enter the IP Address of the device.

Refer to Table 1. Netmask Enter the Netmask to use for the device. Gateway Enter the Gateway to use for the device. DNS Address 1 Enter the DNS server to use for the device. DNS Address 2 Leave blank. Domain Name Enter the network domain name to use.

The sample configuration uses “dr.avaya.com”. Click the Save button to save the network configuration. The screen below shows the configuration used from a device in the sample configuration.




To configure the SIP Global Settings select the Settings option on the left hand side of the screen.

Within the Settings menu select Administrator Options.




Within Administrator Options you will be prompted to input an Administrator password. Input the password and press OK.

To configure the SIP Settings of the Avaya Desktop Video Device within Administrator Options, scroll down to the SIP Settings option.




Within the SIP Settings menu, select the SIP Global Settings option.

Within the SIP Global Settings Option, select the SIP Domain option.




The SIP Domain was set as tandberg.avaya.com. Press the Save button to save the configuration.

In the SIP Settings menu, select SIP Proxy Settings.




Enter the appropriate SIP Proxy settings for the following fields:

SIP Proxy Server Enter the IP Address of the Session Manager: SIP. Refer to Table 1.

Transport Select the transport method. The sample configuration uses “TCP”.

SIP Proxy Port Enter the SIP Proxy Port to use. The sample configuration uses “5060”.

Click the Save button to save the configuration.

3.5.1. Enable G.722 Codec on ADVD The ADVD will download the Axxxsettings.txt configuration file upon boot up. This file is similar to the 46xxsettings.txt file used for the Avaya 46xx and 96xx phone configuration settings. On the http/https server used to store the configuration setting files for the phones, find the Axxxsettings.txt file and open with a text editor. Set the following variables as follows:

SET ENABLE_G722 1 SET ENABLE_G726 0




Note: It was found during testing that both settings, G.722 and G.726, needed to be configured as described above in order to have the ADVD interoperate with the Cisco E20 and other devices using the G.722 codec.

3.6. Administer Cisco VCS These Application Notes assume that the initial network configuration for the Cisco VCS has been completed and the VCS web admin page can be accessed from a computer’s browser on the network. The sample configuration uses IP network settings for the VCS listed in Table 2. Please see Reference 15 in Section 6 for more information about getting started with the VCS. The remaining configuration for the VCS will be done through a web based admin. These configuration areas include:

VCS Configuration o Protocols

H.323 SIP Interworking

o Authentication Devices – Configuration Devices – Local Database

o Zones Default Zone Codian

o Dial plan Configuration Transforms Search Rules

o Bandwidth Configuration




3.6.1. Continuing Configuration Using Web Interface Open a web browser; enter http://xxx.xxx.xxx.xxx for the URL, where xxx.xxx.xxx.xxx is the IP address of the Cisco VCS. Refer to Table 2 for IP Address information of Cisco devices. Click on the Administrator Login button. The default Administrator Login uses “admin” for the Username and “CISCO” for the password. Note: This may be different if the password was changed while following the Cisco Getting Started Guide, Reference 15 in Section 6. After completing the fields with the proper credentials, click the Logon button.

3.6.2. VCS Configuration

3.6.2.1 Protocols The VCS supports the H.323 protocol: it is an H.323 gatekeeper. It will also provide interworking between H.323 and SIP, translating between the two protocols to enable endpoints that only support one of these protocols to call each other. In order to support H.323, the H.323 mode must be enabled.

3.6.2.1.1 Configure H.323 As an H.323 gatekeeper, the VCS accepts registrations from H.323 endpoints and provides call control functions such as address translation and admission control. To configure H.323, start by bringing up the H.323 configuration page by selecting form the top menu VCS configuration → Protocols → H.323. Configure the following fields and leaving any additional fields at their default values:

H.323 mode Select “On”. Registration UDP port Enter the listening port for H.323 UDP registrations.

Default is “1719”. Registration conflict

mode Select “Reject”. Call signaling TCP

port Enter the listening port for H.323 call signaling. Default is “1720”.

Call signaling port range start Enter the numeric value specifying the lower port in the




range used by H.323 calls after they are established. Default is “15000”.

Call signaling port range stop Enter the numeric value specifying the upper port in the range uses by H.323 calls after they are established. Default is “19999”.

Time to live Enter the time interval in seconds for the H.323 endpoints to reregister with the VCS. Default is “1800”.

Call time to live Enter the time interval in seconds the VCS will poll H.323 endpoints in active calls for a response, otherwise disconnect call. Default is “120”.

Auto discover Select if the VCS will respond to “Gatekeeper Discovery Requests” sent out by H.323 endpoints. Default is “On”. The sample configuration uses “Off”.

Caller ID Select if Caller ID displayed on the H.323 endpoint will include or exclude the prefix of the ISDN gateway when displaying the callers E.164 number. Default is “Exclude prefix”. The sample configuration uses “Include prefix”.

Click the Save button to save the configuration. The screen below shows the settings for the sample configuration.




3.6.2.1.2 Configure SIP The VCS supports the SIP protocol. It can act as a:

SIP registrar SIP proxy SIP Presence Server

The VCS can provide interworking between SIP and H.323, translating between the two protocols to enable endpoints that only support one of these protocols to call each other. Even though SIP endpoints were not registered to the VCS in these application notes, the configuration for the SIP protocol had been configured in support of the SIP domain used by the SIP trunk the MCU 4501 uses to connect to the Avaya Aura® Session Manager. Select VCS configuration → Protocols → SIP → Configuration to enter the SIP configuration page. The following fields are configured as described below with all remaining fields left at their default values:

SIP mode To enable SIP functionality select “On”. Registration expire

delta (seconds) Enter the interval in seconds that the SIP endpoints must register and reregister with the VCS. Default is “60”.

SIP registration proxy mode Enter how proxied registrations and requests containing Route Sets are handled: Off, Proxy to known only, or Proxy to any. Default if “Off”. The sample configuration sets this field to “Proxy to any”.

UDP mode Select “On” to enable UDP. UDP port Select “5060” as the listen port for UDP. TCP mode Select “On” to enable TCP. TCP port Select “5060” as the listen port for TCP. TLS mode Select “On” to enable TLS. TLS port Select “5061” as the listen port for TLS. TCP outbound

port start Specify the lower port in the range to use for TCP outbound when TCP connections are established. Default is “25000”.

TCP outbound port end Specify the upper port in the range to use for TCP




outbound when TCP connections are established. Default is “29999”.

Session refresh interval (seconds) Enter “1800”.

Minimum session refresh interval (secs) Enter “180”.

Require UDP BFCP mode Select “On”. Default is “On”.

Require duo video mode Select “On”. Default is “On”.

Click the Save button to save the configuration. The screen below shows the settings for the sample configuration.

Add a SIP Domain by selecting VCS configuration → Protocols → SIP → Domain and clicking the New button. Enter the SIP Domain to be used and click on the Create domain button. The sample configuration uses “tandberg.avaya.com”.




3.6.2.1.3 Interworking To enable interworking between H.323 and SIP select VCS configuration → Protocols → Interworking. In the “H.323 <-> SIP interworking mode” field, select “On”. Click the save button to save the configuration.

3.6.2.2 Authentication In order for an endpoint to use the VCS as its H.323 gatekeeper or SIP registrar, the endpoint must first register with the VCS. The VCS can be configured to control which devices are allowed to register with it. Two separate mechanisms are provided:

a device authentication process based on the username and password supplies by the endpoint

a simple Registration Restriction Policy that uses Allow Lists or Deny Lists to specify which aliases can and cannot register with the VCS, and the ability to control registrations based on IP addresses and subnet ranges through the specification of subzone membership rules and subzone registration policies1

The sample configuration uses the first method of username and password authentication provided by the endpoint to the VCS.

1 Note. From Cisco TelePresence Video Communication Server Administration Guide, Software Version X6, Doc ID: D14049.09, December 2010




3.6.2.2.1 Devices – Configuration Select VCS configuration → Authentication → Devices → Configuration to enter the configuration page. In the “Database type” field select “Local database” and then click the Save button.

3.6.2.2.2 Devices – Local Database Select VCS configuration → Authentication → Devices → Local Database. To add a new entry, click the New button. Refer to Table 4 and enter the Name of the first device and a desired password. Click Create credential to enter the device into the local database. The following screen below shows an entry from the sample configuration.

Repeat for each of the H.323 devices listed in Table 4.

3.6.2.3 Zones

3.6.2.3.1 DefaultZone By default the “DefaultZone” is automatically created. To check the settings of the “DefaultZone” select VCS configuration → Zones → DefaultZone. Set the field for “Authentication policy” to “Check credentials” and click on the Save button.




3.6.2.3.2 Codian Create a new zone between the VCS and the MCU 4501. Select VCS configuration → Zones and click on the New button. The following fields are configured as described below with all remaining fields left at their default values: Under the Configuration section:

Name Enter a descriptive name for the zone. The sample configuration uses “Codian”.

Type Select “Neighbor”. Hop count Enter “15”. Default is “15”.

Under the H.323 section:

Mode Select “On”. Port Enter “1719”.

Under the SIP section:

Mode Select “Off”. Port Enter “5060”. Transport Select “TCP”. Accept proxied

registrations Select “Deny”. Under the Authentication section:




Authentication Policy Select “Check credentials”.

Under the Location section:

Peer 1 address Enter IP Address of MCU 4501 from Table 2. Under the Advanced section:

Zone profile Select “Default”. Click the Save button to save configuration. The sample configuration for the “Codian” zone is shown below.




3.6.2.4 Dial plan

3.6.2.4.1 Configuration Select VCS configuration → Dial plan → Configuration and select “Direct” for the field “Calls to unknown IP addresses”. Click the Save button.

3.6.2.4.2 Transforms Select VCS configuration → Dial plan → Transforms. Click on the New button and enter the following field information, leaving the remaining fields set at their default values:

Priority Enter the priority level for the transform. The sample configuration uses “1”.

Description Enter a brief description of the transform. The sample configuration uses “Cisco H323 to H323”.

Pattern type Select how the pattern string must match the alias for the transform to be applied. The selections are: Exact, Prefix, Suffix, and Regex. The sample configuration uses “Regex”.

Pattern string Enter the pattern against which the alias is compared. The sample configuration uses “(65\d+)”.

Pattern behavior Select how the alias is modified. The sample configuration uses “Replace”.

Replace string Enter the text string to use in conjunction with the selected Pattern behavior. The sample configuration uses “\1”.

State Select the state of the transform, Enabled or Disabled. The sample configuration uses “Enabled”.

Click the Create Transform button to save the new transform configuration. Note: the sample configuration screen below was taken from the edit page and shows the Save button instead.




Repeat the steps above to create transforms for the list shown below.

3.6.2.4.3 Search Rules Select VCS configuration → Dial plan → Search rules. Click the New button to enter the search rule configuration page. Enter the following field values and leaving the remaining fields set to their default values:

Rule name Enter a name for the new rule. The sample configuration uses “Codian”.

Description Enter a short description of the rule. Priority Enter the priority of the search rule with the highest priority

starting at 1 and going up to 65534. The default is “100”. The sample configuration uses “99”.

Source Select the source of the requests for which this rule is applied. The sample configuration uses “Any”.

Request must be authenticated Specifies whether this search rule applies only to authenticated search requests. The sample configuration uses “No”.




Mode Select the type of alias for which this search rule applies. The sample configuration uses “Alias pattern match”.

Pattern type Select how the pattern string must match the alias for the rule to be applied. The sample configuration uses “Prefix”.

Pattern string Enter the pattern against which the alias is compared. The sample configuration uses “695”.

Pattern behavior Select how the matching part of the pattern is modified. The sample configuration uses “Leave”.

On successful match Select the ongoing search behavior if the alias matches all of the search rule’s conditions. The sample configuration uses “Continue”.

Target Select the zone or policy service to query if the alias matches the search rule. The sample configuration uses “Codian”.

State Select if the search rule is enabled or disabled. The sample configuration uses “Enabled”.

Click the Create search rule button to save the new search rule. Note: The sample configuration screen below was taken from the edit page and shows the Save button instead.




3.6.2.5 Bandwidth

3.6.2.5.1 Configuration Increase the default call bandwidth by selecting VCS configuration → Bandwidth → Configuration to enter the configuration page. Enter the following field values:

Default call bandwidth (kbps) Enter “5120”. The default is “384”.

Downspeed per call mode Select “On”.

Downspeed total mode Select “On”.

Click the Save button.

3.7. Administer Cisco TelePresence MCU 4501 These Application Notes assume that the initial network configuration for the Cisco TelePresence MCU 4501 has been completed and the MCU 4501 web admin page can be accessed from a computer’s browser on the network. The sample configuration uses IP network settings for the MCU 4501 listed in Table 2. Please see Reference 18, in Section 6 for more information about getting started with the MCU 4501 4500 series. The remaining configuration for the MCU 4501 will be done through a web based admin. These configuration areas include:

Network o Services

Settings o Conference Settings o H.323 Gatekeeper Settings




o SIP Registrar Settings o Streaming Settings o Content Settings o Media Ports o Time NTP Settings o Security Settings

Endpoints o H.323 o SIP

Conferences o Auto Attendant o Creating a Conference

3.7.1. Continue Configuration Using Web Interface Open a web browser, enter http://xxx.xxx.xxx.xxx for the URL, where xxx.xxx.xxx.xxx is the IP address of the Tandber/Cisco TelePresence MCU 4501 . Refer to Table 2 for IP Address information of Cisco devices. Click on the Log in button. The default Administrator Login uses “admin” for the Username and leave the Password field blank. Note: This may be different if the password was changed while following the Cisco Getting Started Guild for MCU 4500 Series, Reference 18 in Section 6. After completing the fields with the proper credentials, click on the OK button.




3.7.2. Administer Network Services Select Network → Services tab to enter the Services configuration page. Check the following settings enabling the service on Port A and configuring the correct port number to use for that service. Under TCP service:

Web Enabled Port: 80 Incoming H.323 Enabled Port: 1720 Incoming SIP (TCP) Enabled Port: 5060 Streaming (Windows

Media Player) Enabled Port: 1755 Streaming (other) Enabled Port: 554 FTP Enabled Port:21 ConferenceMe Unchecked .




Under UDP service:

SNMP Disabled Port: 161 SIP (UDP) Enabled Port: 5060 H.323 gatekeeper Enabled Port: 1719 Tunneled media Disabled Port: 5082

Click the Apply changes button to save the configuration. The following screen shows the sample configuration for this configuration section.

3.7.3. Administer Settings

3.7.3.1 Conference Settings Select Settings → Conferences tab to enter the Conference settings configuration page. Enter the following field values and leave the remaining fields set to their default values: Under Conference Settings:

Maximum video size Select “Receive MAX, transmit MAX”. Motion / sharpness tradeoff Select “Favor sharpness”. Transmitted video resolutions Select “Allow all resolutions”. Default bandwidth from MCU Select “4.00 Mbits/s” Default bandwidth to MCU Select “2.00 Mbits/s”




Default view family Select “2x2, 3x3, or 4x4 as appropriate”. Use full screen view for

two participants Select “Disabled”. Active speaker display Select “Green border”. Media port reservation Select “Disabled”.

Audio notifications Check only the following:

Conference timing Conference status Join and leave indications

Overlaid icons Check only the following: Important participant Unsecured conferences Tunneled camera control Layout changes

Overlaid text Check only the following: Conference status Confernece timing Join and leave indications Text messages Content channel text chat

Overlaid logo duration Select “5 seconds”. Time to show participant names Select “5 seconds”.




Under Advanced settings:

Audio codecs from MCU Check only the following: G.711 G.722 G.722.1 G.729 G.723.1 Polycom® Siren14™ G.722.1 Annex C AAC-LD AAC-LC

Audio codecs to MCU Check only the following: G.711 G.722 G.722.1 G.729 G.723.1 Polycom® Siren14™ G.722.1 Annex C AAC-LD AAC-LC

Video codecs from MCU Check only the following: H.261 H.263 H.263+ H.263 interlaced H.264

Video codecs to MCU Check only the following: H.261 H.263 H.263+ H.263 interlaced H.264

ClearVision Checked . Video transmit size optimization Select “Dynamic codec and resolution”. Video resolution selection mode Select “Default”. Video format Select “NTSC – 30 fps”.




Maximum transmitted video packet size Enter “1400” bytes.

Interlaced video optimization Unchecked . Don’t see yourself in small panes Unchecked . Don’t duplicate in small panes Unchecked . Automatically make content

channel important Unchecked . Loudest speaker pane placement

behavior Select “Allow duplication of placed participants in small panes only”.

Pane rolling interval Select “5 seconds”. Voice switching sensitivity Enter “50” % Additional audio delay relative

to video Select “0 ms”. Floor and chair control Select “Disabled”. Note: this disables

BFCP by default for all conferences created. Incoming calls to unknown

conference or auto attendants Select “Default auto attendant”. Failed preconfigured participants

redial behavior Select “Default”. Conferences remain locked

when empty Unchecked . Use conference name as caller ID Unchecked . Require H.323 gatekeeper callers

to enter PIN Unchecked . Time to wait when setting up ad hoc

conference PIN Select “10 seconds”. Advertise out of band DTMF

to endpoints Checked . Click the Apply changes button to save configuration.




3.7.3.2 H.323 Gatekeeper Settings Select Settings → H.323 tab to enter the H.323 gatekeeper settings configuration page. Enter the following field values and leave the remaining fields set to their default values:

H.323 gatekeeper usage Select “Enabled”. H.323 gatekeeper address Enter the IP Address of the VCS.

Refer to Table 2. Gatekeeper registration type Select “MCU (standard)”. Ethernet port association Check Port A. (Mandatory) H.323 ID

to register Enter the extension assigned to the MCU 4501 MCU from Table 4. The sample configuration uses “69500”.

Use Password Checked . Password Enter the matching password used to create

the device for the MCU 4501 on the VCS in Section 3.6.2.2.2.

Prefix for MCU registrations Leave blank. MCU service prefix Enter the extension number of the MCU

4501 MCU. Allow numeric ID registration

for conferences Checked . Send resource availability indications Unchecked . Thresholds: conferences Leave blank. Threshold: video ports Leave blank.

Click the Apply changes button to save configuration.




3.7.3.3 SIP Registrar Settings Select Settings → SIP tab to enter the SIP registrar settings configuration page. Enter the following field values and leave the remaining fields set to their default values: Under SIP registrar settings section:

SIP registrar usage Select “Enabled”. SIP registrar domain Enter the IP Address of the Session

Manager: SIP. Refer to Table 1. SIP registrar type Select “Standard SIP”. Username Enter the extension number of the MCU

4501 MCU. This will be the username created in Section 3.1.12.

Password Enter the password of the above user. Allow number ID registration

for conferences Unchecked . Under the SIP call settings section:

SIP proxy address Enter the IP address of the Session Manager: SIP from Table 1.

Maximum bit rate from Microsoft OCS/LCS clients Select “1.00 Mbit/s”.

Outgoing Transport Select “TCP”. Use local certificate for outgoing

connections and registrations Unchecked . Click the Apply changes button to save the configuration.




3.7.3.4 Streaming Settings Select Settings → Streaming tab to enter the Streaming & ConferenceMe settings configuration page. Enter the following field values and leave the remaining fields set to their default values: Under the Streaming & ConferenceMe settings section:

Enable Select “Streaming”. Under the Streaming settings section: Codec and bit rate choice 1:

Name Leave blank. Player format Select “Windows Media Player”. Streams Select “256k, Polycom® Siren14™ audio,

H.264 video”. Codec and bit rate choice 2:

Name Leave blank. Player format Select “Windows Media Player”. Streams Select ”384k, Polycom® Siren14™ audio,

H.264 video”. Multicast:

Media transmit IP address range Leave range blank.

Media tranmit port

number range Leave range blank.

Windows Media Player: Protocol Select “HTTP”.

Under the ConferenceMe settings section:

Allow fall back to media using TCP Unchecked .

Maximum ConferenceMe connections Select “3”.

Maximum bandwidth from MCU Select “768 kbit/s”.

Maximum bandwidth to MCU Select “768 kbit/s”.

Click the Apply changes button to save configuration.




The screen below shows the settings for streaming used by the sample configuration.

3.7.3.5 Content Settings Select Settings → Content tab to enter the Content settings configuration page. Enter the following field values and leave the remaining fields set to their default values:

Content status Select “Disabled”. Display content in normal

video channel Select “Enabled”. Video fast update request

filtering Select “Enabled”. Web viewing applet

bandwidth Select “384 kbit/s”.




Markup of content channel video Select “Disabled”.

Automatic content handover Select “Disabled”. Click the Apply changes button to save configuration. The screen below shows the settings for content used by the sample configuration.




3.7.3.6 Media Ports Select Settings → Media ports tab to enter the Media port allocation configuration page. Enter the following field values and leave the remaining fields set to their default values:

Media port mode Select “HD”. Click the Apply changes button to save configuration.

3.7.3.7 Time (NTP) Settings Select Settings → Time tab to enter the Time settings configuration page. Enter the following field values and leave the remaining fields set to their default values: Under the NTP section:

Enable NTP Checked . UTC offset Enter the Universal Time Clock offset.

The sample configuration uses “0”. NTP host Enter the IP Address of the NTP server.

The sample configuration uses “192.168.1.2”. Click the Update NTP settings button to save configuration.




3.7.3.8 Security Settings Select Settings → Security tab to enter the Security settings configuration page. Enter the following field values and leave the remaining fields set to their default values: Under the Security settings section:

Advanced account security mode Select “Disabled”.

Idle web session timeout Enter “60” minutes. Click the Update security settings button to save configuration. Under the Serial console settings section:

Hide log messages on console Unchecked . Disable serial input during

startup Unchecked . Require administrator login Unchecked .

Click the Update console settings button to save configuration. The screen below shows the security settings used by the sample configuration.




3.7.4. Administer Endpoints Endpoints are added to the MCU 4501 to allow conferences to be able to add participants for the dial-out feature. This allows a conference which is setup with dial-out participants to call the endpoints extensions when the conference starts. The sample configuration for the MCU 4501 contains both Cisco (H.323) endpoints and Avaya (SIP) endpoints. This section will describe how to configure an endpoint on the MCU 4501 and will refer to both Table 3 and Table 4 for names and extensions. Select Endpoints from the top menu selection to enter the Endpoints tab which lists all created endpoints.

3.7.4.1 H.323 Endpoint To create an H.323 endpoint, click the Add H.323 button. Enter the following field values and leave the remaining fields set to their default values: Display parameters:

Name Enter the name associated with the endpoint. Refer to Table 4.

Call-out parameters: Address Enter the IP Address, host name, or an E.164

address (phone number). H.323 gateway Select “<none>”. DTMF sequence Leave blank.

Call-in match parameters:

Name Enter the name associated with the endpoint. IP Address Leave blank. E.164 Enter the extension number. Refer to Table 4.

Conferencing parameters:

Display name override The name that will be displayed in a conference as a label for this endpoint. Leave blank.

Motion / sharpness tradeoff Select “<use box-wide setting>”. Transmitted video resolutions Select “<use box-wide setting>”. Content contribution Select “<use conference configuration>”. Content receive Select “Enabled”. View border size Select “<no border>”. Default view family Select “<use box-wide setting>”. Preferred bandwidth

from MCU Select “<use box-wide setting>”. Preferred bandwidth

to MCU Select “<use box-wide setting>”.




Layout control via FECC / DTMF Select ”<use conference configuration>”.

Mute in-band DTMF Select “<use conference configuration>”. Initial audio status Select “Active”. Initial video status Select “Active”. Automatic disconnection Select “Disabled”. Custom codec selection Select “Disabled”.

Click Add endpoint button to add the new endpoint. The screen below shows an example H.323 endpoint configuration from the sample configuration.




3.7.4.2 SIP Endpoint To create a SIP endpoint, click the Add SIP button. Enter the following field values and leave the remaining fields set to their default values: Display parameters:

Name Enter the name associated with the endpoint. Refer to Table 3.

Call-out parameters: Address Enter the IP Address, host name, or an E.164

address (phone number). Use SIP registrar Checked . Outgoing transport Select “<use box-wide setting>”. DTMF sequence Leave blank.

Call-in match parameters:

Username Enter the name associated with the endpoint. IP Address Leave blank.

Conferencing parameters:

Display name override The name that will be displayed in a conference as a label for this endpoint. Leave blank.

Motion / sharpness tradeoff Select “<use box-wide setting>”. Transmitted video resolutions Select “<use box-wide setting>”. Content contribution Select “<use conference configuration>”. Content receive Select “Enabled”. View border size Select “<no border>”. Default view family Select “<use box-wide setting>”. Preferred bandwidth

from MCU Select “<use box-wide setting>”. Preferred bandwidth

to MCU Select “<use box-wide setting>”. Layout control via

FECC / DTMF Select ”<use conference configuration>”. Mute in-band DTMF Select “<use conference configuration>”. Initial audio status Select “Active”. Initial video status Select “Active”. Automatic disconnection Select “Disabled”. Custom codec selection Select “Disabled”.

Click Add endpoint button to add the new endpoint. The screen below shows an example SIP endpoint configuration from the sample configuration.




3.7.5. Administer Conferences

3.7.5.1 Configure Auto Attendant Select Conference → Auto attendant tab and select Default auto attendant. Enter the following field values and leave the remaining fields set to their default values: Under the Auto attendant section:

Name Select “Default”. Title Leave blank. Numeric ID Leave blank. Numeric ID registration Unchecked H.323 gatekeeper

Unchecked SIP registrar PIN Leave blank. Re-enter PIN Leave blank. Creation of new conferences Select “Enabled”. Access to ad hoc conferences Select “Enabled”.

Under the Accessible scheduled conferences section:

All scheduled conferences Radio button selected . Click the Update auto attendant button to save configuration. The screen below shows the default auto attendant configuration from the sample configuration.




3.7.5.2 Creating a Conference

3.7.5.2.1 Ad-hoc Conferences Created by the Auto Attendant Endpoints can create an Ad-hoc conference by dialing into the MCU 4501 using the assigned extension, in the sample configuration the conference server extension is “69500”. The video endpoint will be presented with a welcome banner and voice instructions on joining or creating a conference.

A new conference can be created by pressing the “*” key on the calling endpoints keypad. A prompt for the conference ID will be presented. Enter a numeric number to use as the conference ID others will use to join the conference, followed by the “#” key.

The next prompt will be to enter the new conference PIN. Enter a numeric number to use as the conference PIN others will use to join the conference, followed by the “#” key.




The endpoint who is the conference creator will be the first person to join the conference. Other audio/video endpoints can join the newly created conference by dialing the conference server (69500), entering the conference ID number followed by the ”#” and then entering the conference PIN followed by the “#”.

3.7.5.2.2 Scheduled Conference from Web Admin A scheduled conference can be created from the web administration page to start at a specific date/time and also if needed have the conference dial-out to the conference participants. Select Conferences → Conferences tab. Under the “No scheduled conferences” section, click the “Add conference” button to enter configure mode. Enter the following field values and leave the remaining fields set to their default values: Under the Parameters section:

Name Enter the name that users will see on auto attendant screens and on the MCU’s web interface.

Description A short description of the conference. Numeric ID The numeric identifier of the conference. This is

the number conference participants will use at the auto attendant screen to join the conference.

PIN The PIN used to join the conference is set. Guest numeric ID Leave blank. Guest PIN Leave blank. Owner Select “admin”. Numeric ID registration Unchecked H.323 gatekeeper

Unchecked SIP registrar When only guests remain Select ”Disconnect all participants”. Floor and chair control Note: Disabled on Conference settings page. Automatic lecture mode Select “Disabled”.




Visibility Select “Public”. Layout control via

FECC / DTMF Select “FECC with DTMF fallback”. Mute in-band DTMF Select “When used for MCU control”. Allow DTMF *6 to mute audio Select “Enabled”. Invite pre-configured

participants Select “At the start of conference”. Mute on join Unchecked Audio

Unchecked Video Streaming Select “Unicast”. Content channel video Select “Enabled”. Content contribution

from endpoints: Select “Enabled”. Transmitted content

resolutions Select “Allow all resolutions”. Maximum video participants Leave blank. Maximum audio-only

participants Leave blank. Under the Start time and duration section:

Start time Enter the start time for the conference. Start date Enter the start date. Permanent Unchecked . Maximum duration Set duration in days, hours, minutes.

Under the Repetition section:

Interval Select radio button for “No repeat”. Termination Select radio button for “No end date”.

To add endpoints for the conference to dial-out to when the conference starts, click the Pre-configured participants button. Check the box next to all participants who are to attend the conference. And click the Return to conference configuration button. Click the Add conference button to add the conference. The screen below shows a sample conference with participants added for dial-out.




3.8. Administer Cisco C20 Video Endpoint These Application Notes assume that the initial network configuration for the Cisco Quick Set C20 has been completed and the C20 web admin page can be accessed from a computer’s browser on the network. The sample configuration uses IP network settings for the C20’s listed in Table 2. Please see Reference 23 & 24 in Section 6 for more information about getting started with the C20. The remaining configuration for the C20 will be done through a web based admin. These configuration areas include:

Administering H.323 o Profile 1

H323Alias CallSetup Gatekeeper Authentication

Network Services o H323 o NTP

3.8.1. Continuing Configuration Using Web Interface Open a web browser; enter http://xxx.xxx.xxx.xxx for the URL, where xxx.xxx.xxx.xxx is the IP address of the Cisco C20. Refer to Table 2 for IP Address information of Cisco devices. The default Administrator Login uses “admin” for the Username and “CISCO” for the password. Note: This may be different if the password was changed while following the Cisco Administrator’s Guide for the C20, Reference 24 in Section 6. After completing the fields with the proper credentials, click the Sign In button.




3.8.2. Administering H.323 Select Advanced Configuration from the left menu. Expand the H323 folder. Now expand each of the following folders:


Enter the following field values and leave the remaining fields set to their default values. Under the H323Alias section:

ID: Enter a numeric extension number. Refer to Table 4. The sample configuration uses “65000”.

E164: Enter the E.164 number for this endpoint. Under the CallSetup section:

Mode: Select “Gatekeeper”. Under the Gatekeeper section:

Address: Enter the IP Address of the VCS acting as a gatekeeper. Refer to Table 2.

Discovery Select “Manual”. Under the Authentication section:

LoginName: Enter the extension number of the endpoint. The sample configuration uses “65000”.

Password: Enter the password for this endpoint. Mode: Select “On”.

The screen below shows the configuration used by a C20 from the sample configuration.




3.8.3. Administer Network Services Select Advanced Configuration from the left menu. Expand the NetworkServices folder and the H323 and SIP sub-folders. Select “On” mode for the H323 and “Off” for the SIP. Now expand the NTP folder. Enter for following field values:

Mode: Select “Manual”. Address: Enter the Network Time Server’s IP Address.

The screen below shows the configuration used by a C20 from the sample configuration.




3.9. Administer Cisco 1700 MXP Video Endpoint These Application Notes assume that the initial network configuration for the Cisco 1700 MXP has been completed and the 1700 MXP web admin page can be accessed from a computer’s browser on the network. The sample configuration uses IP network settings for the 1700 MXP listed in Table 2. Please see Reference 22 in Section 6 for more information about getting started with the 1700 MXP. The remaining configuration for the 1700 MXP will be done through a web based admin. These configuration areas include:

System Configuration o Administering H323 Settings

H323 Settings Gatekeeper Settings

3.9.1. Continuing Configuration Using Web Interface Open a web browser; enter http://xxx.xxx.xxx.xxx for the URL, where xxx.xxx.xxx.xxx is the IP address of the Cisco 1700 MXP. Refer to Table 2 for IP Address information of Cisco devices. The default Administrator Login uses “admin” for the Username and “CISCO” for the password. Note: This may be different if the password was changed while following the Cisco Administrator’s Guide for the 1700 MXP, Reference 22 in Section 6. After completing the fields with the proper credentials, click the Ok button. The “Welcome” menu screen appears.




3.9.2. Administer System Configuration Click on the System Configuration tab at the top of the menu.

3.9.2.1 H323 Settings Click the H.323 menu item under the System Configuration tab. Enter the following field values and leave the remaining fields set to their default values: Under the H323 Settings section:

Callsetup Mode Select “Gatekeeper”. H.323 Auto Prefix Leave blank. E.164 Alias Enter the extension number of the device.

The sample configuration uses “65005”. Under the Gatekeeper Settings section:

H.323 ID Enter the extension number of the device. Gatekeeper Discovery Select “Manual”. Gatekeeper Address Enter the IP Address of the VCS (gatekeeper).

Refer to Table 2. Authentication Mode Select “Auto”. Authentication ID Enter the extension number of the device. Authentication Password Enter the password of the device. Multiple H.323 Aliases Select ”Off”.

Click the Save button to save the configuration.




3.10. Administer Cisco EX-90 Video Endpoint These Application Notes assume that the initial network configuration for the Cisco EX-90 has been completed and the EX-90’s web admin page can be accessed from a computer’s browser on the network. The sample configuration uses IP network settings for the EX-90 listed in Table 2. Please see References 20 and 21 in Section 6 for more information about getting started with the EX-90. The remaining configuration for the EX-90 will be done through a web based admin. These configuration areas include:

Administering H.323 o Profile 1


Network Services o H323 o NTP

3.10.1. Continuing Configuration Using Web Interface Open a web browser; enter http://xxx.xxx.xxx.xxx for the URL, where xxx.xxx.xxx.xxx is the IP address of the Cisco EX-90. Refer to Table 2 for IP Address information of Cisco devices. The default Administrator Login uses “admin” for the Username and “CISCO” for the password. Note: This may be different if the password was changed while following the Cisco Administrator’s Guide for the EX-90, Reference 20 in Section 6. After completing the fields with the proper credentials, click the Sign In button.




3.10.2. Administering H.323 Select Advanced Configuration from the left menu. Expand the H323 folder and then expand the Profile 1 folder. Now expand each of the following folders:


Enter the following field values and leave the remaining fields set to their default values. Under the H323Alias section:

ID: Enter a numeric extension number. Refer to Table 4. The sample configuration uses “65010”.

E164: Enter the E.164 number for this endpoint. Under the CallSetup section:

Mode: Select “Gatekeeper”. Under the Gatekeeper section:

Address: Enter the IP Address of the VCS acting as a gatekeeper. Refer to Table 2.

Discovery Select “Manual”. Under the Authentication section:

LoginName: Enter the extension number of the endpoint. The sample configuration uses “65010”.

Password: Enter the password for this endpoint. Mode: Select “On”.

The screen below shows the configuration used by the EX-90 from the sample configuration.




3.10.3. Administer Network Services Select Advanced Configuration from the left menu. Expand the NetworkServices folder and the H323 and SIP sub-folders. Select “On” mode for the H323 and “Off” for the SIP. Now expand the NTP folder. Enter for following field values:

Mode: Select “Manual”. Address: Enter the Network Time Server’s IP Address.

The screen below shows the configuration used by the EX-90 from the sample configuration.




4. Verification The following verification steps were tested using the sample configurations. The following steps can be used to verify portions of the installation in the field.

1. Verified the Avaya Desktop Video Device extension 60010 was registered to the Session Manager. Verified the extension 60010 was logged in successfully to the Avaya Desktop Video Device.

2. Verified the Avaya 10x0 series endpoints were registered to the Session Manager. Verified the configured extensions for the Avaya 10x0 series were logged in successfully to their respective Avaya 10x0 series units.

3. Verified SIP Entity Link between the Session Manager and the Cisco TelePresence MCU 4501 was up and active.

4. Verified the Cisco video endpoints were registered with the VCS as their H.323 gatekeeper.

5. Verified each of the Cisco video extensions were logged into their respective video units.

6. Verified MCU 4501 was registered as a H.323 endpoint on the VCS.

7. Verified Cisco H.323 video endpoints could dial into the MCU 4501 conference server and establish a new conference.

8. Verified Avaya SIP endpoints could dial into the MCU 4501 via SIP trunk between the Session Manager and MCU 4501 to establish video conferences and join others.

9. Verified two-way video and audio in all conference calls by Cisco H.323 endpoints and Avaya SIP endpoints.

10. Verified dial-out from MCU 4501 conference to both Cisco H.323 video endpoints and Avaya SIP video endpoints.

11. Verified testing of people plus content with BFCP disabled on MCU 4501 conference server to make sure no adverse effects were noticed.

12. Verified Avaya one-X® Communicator in both H.323 and SIP modes could dial into the MCU 4501 via SIP trunk between the Session Manager and MCU 4501 to establish video conferences and join others.




4.1. Avaya HQ

4.1.1. Registered Users on Avaya Aura® Session Manager Access Session Manager → System Status → User Registrations to see the Avaya endpoints registered to Session Manager.

4.1.2. Avaya Aura® Session Manager Entity Link Status Access Session Manager → System Status → SIP Entity Monitoring to see the SIP Entity Link status between the Session Manager and the MCU 4501.




4.2. Cisco Remote Branch / B2B Partner

4.2.1. Registered Users on VCS Access the VCS and select Status → Registrations → By Device to list all the devices registered to the VCS.

4.2.2. MCU 4501 Conference Server Access the MCU 4501 and select Settings → SIP to display the SIP registrar settings and see if the conference server is registered with the Session Manager.




5. Conclusion These Application Notes have described the administration steps required to configure Avaya Aura® Session Manager with Avaya Aura® Communication Manager running as an Evolution Server, Avaya 10x0 series video endpoints, Avaya one-X® Communicator, and Avaya Desktop Video Devices registered via SIP to the Session Manager. Also included in these Application Notes is the configuration of Cisco’s Video Communications Server (VCS) and MCU 4501 conference server supporting Cisco video endpoints (C20, 1700 MXP, and EX-90) using the H.323 protocol locally in a remote site location. A SIP trunk configured between the Avaya Aura® Session Manager and the Cisco TelePresence MCU 4501 allows conferencing interoperability between both Avaya video supported endpoints and Cisco video supported endpoints.




6. References The following references are relevant to these Application Notes: Avaya

[1] Avaya Aura® Session Manager Overview, Doc ID 03-603323, available at http://support.avaya.com.

[2] Installing and Administering Avaya Aura® Session Manager, Doc ID 03-603324, available at http://support.avaya.com.

[3] Avaya Aura® Session Manager Case Studies, dated January 2, 2010, available at http://support.avaya.com

[4] Maintaining and Troubleshooting Avaya Aura® Session Manager, Doc ID 03-603325, available at http://support.avaya.com.

[5] Hardware Description and Reference for Avaya Aura® Communication Manager (COMCODE 555-245-207) http://support.avaya.com/elmodocs2/comm_mgr/r4_0/avayadoc/03_300151_6/245207_6/245207_6.pdf

[6] SIP Support in Avaya Aura® Communication Manager Running on Avaya S8xxx Servers, Doc ID 555-245-206, May 2009, available at http://support.avaya.com.

[7] Administering Avaya Aura® Communication Manager, Doc ID 03-300509, May 2009, available at http://support.avaya.com.

[8] Avaya 1010/1020 Installation Guide, Issue 1, June 2010, available at http://support.avaya.com

[9] Avaya 1010/1020 User Guide, Issue 1, June 2010, available at http://support.avaya.com

[10] Avaya 1030 Installation Guide, Issue 1, June 2010, available at http://support.avaya.com






[13] Avaya Video Communications System Administrator Guide (1050/1040/1030) , Issue 1, June 2010, available at http://support.avaya.com

[14] Avaya one-X Communicator Getting Started R6, August 2010, available at http://support.avaya.com

Cisco [15] Cisco TelePresence Video Communication Server: Getting Started, Version X6, Doc

ID: D14350.05, December 2010, available at http://www.Cisco.com

[16] Cisco TelePresence Deployment Guide: Single Cisco VCS Control, Doc ID: D14524.02, October 2010, available at http://www.Cisco.com

[17] Cisco TelePresence Video Communication Server Administration Guide, Software Version X6, Doc ID: D14049.09, December 2010, available at http://www.Cisco.com

[18] Cisco TelePresence MCU 4500 Series Getting Started, Doc ID: 61-0011-07, available at http://www.Cisco.com

[19] Connecting H.323 MCUs to Cisco VCSs for resilience and capacity: Cisco TelePresence Deployment Guide, Doc. ID: D14375.02, November 2010, available at http://www.Cisco.com

[20] Cisco TelePresence System EX90 User Guide, Software Version TC4.0, Doc ID: D14724.03, November 2010, available at http://www.Cisco.com

[21] Cisco TelePresence System EX90/EX60 Administrator Guide, Software Version TC4.0, Doc ID: D14726.03, December 2010, available at http://www.Cisco.com

[22] Cisco MXP Administrator guide, Software Version F9, Doc ID: D14033.08, October 2010, available at http://www.Cisco.com

[23] Cisco TelePresence System Profile-series, C-series codecs, Quick Set C20: User Guide, Software Version TC4.0, Doc ID: D14582.04, November 2010, available at http://www.Cisco.com

[24] Cisco TelePresence System Quick Set C20/C20 Plus and Profiles using C20: Administrator Guide, Software Version TC4.0, Doc. ID: D14637.04, November 2010, available at http://www.Cisco.com




©2011 Avaya Inc. All Rights Reserved. Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and ™ are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes. Please e-mail any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya Solution & Interoperability Test Lab at [email protected]

Configuration for Interoperability of Avaya Aura® Session

Documents

Transcript of Configuration for Interoperability of Avaya Aura® Session